Teleconference terminal

ABSTRACT

In the PC-based teleconference terminal 200 includes a PC 110 and a video codec unit 130, a network control unit 220 and an audio codec unit 240 which are all constructed on the same expansion board, with the video codec unit 130, the audio codec unit 240 and the network control unit 240 all connected to the computer bus 114 so as to be able to transfer audio data, video data, data and AV multiframes between themselves. The audio codec unit 240 is equipped with a audio clock generation unit 241 for generating an audio sampling signal of 8 kHz through self-excited oscillation. The CPU 111 in the PC 110 executes frame alignment by executing the AV multiplexer/separator software 212 stored in the memory 112 and executes the AV multiframe conversion and separation for H series recommendation on the CCITT, as well as adjusting any shortages or surpluses of reproduction audio data which arise due to synchronization slips between the audio sampling clock and the network clock by executing the synchronization slip control software 211.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to a teleconference terminal connected toother such terminals located at remote points by a network, which caninter-communicatively execute teleconferencing by transmitting audio andvideo information in a multiplex format using line switching on an ISDNline.

2. Description of the Related Art

In recent years, the use of ISDN lines has become commonplace, and this,as well as the establishment of a CCITT recommendations, such as Hseries, as standard protocols for teleconference transmission, and thereductions in the cost of TV picture codecs have led to the developmentof teleconference terminals, based on widely-used personal computers(PC), which are capable of video telephone services using ISDN networks.This kind of teleconference terminal is known as a PC-basedteleconference terminal.

FIG. 1 shows a construction of an PC-based teleconference terminal 100according to the related art. This PC-based teleconference terminal 100is a terminal for H.320 recommendation on the CCITT, and is principallycomprised of a PC 110 and an AV unit 170. The AV unit 170 is constructedas an single enclosed unit, and includes a specialized network controlunit 120, a video codec unit 130, an audio codec unit 140, a CPUinterface 160, a control bus 161, and a time sharing bus 162. The PC110, the video codec unit 130, and the audio codec unit 140 areconnected to the display unit 150, the camera 151 and the microphone 152and speaker 153, respectively.

The PC 110 is made up of the CPU 111, a memory 112 and a display controlunit 113 and the computer bus 114.

The CPU 111 is composed of a widely-used microprocessor, and executesdata processing using the data received from the ISDN line by executinga program stored in the memory 112.

The memory 112 stores the programs to be executed by the CPU 111, aswell as providing the working-storage for when the CPU 111 is executingdata processing.

The display control unit 113 controls the operation of the display unit150 and has the video signals expanded and decoded by the video codecunit 130 displayed by the display unit 150. The display control unit 113is equipped with a function for having a number of images superimposedand displayed by the display unit 150.

The computer bus 114 provides the data transfer lines for connecting allof the components inside the PC 110 with the CPU interface 160 in the AVunit 170.

The specialized network control unit 120 is comprised of a lineswitching connection processing unit 121, an ISDN line interface unit122, a network clock control unit 123 and an AV multiplexer/separatorunit 124.

The line switching connection processing unit 121, executes theprocesses connecting and disconnecting the line switching on the Dchannel of the ISDN line via the ISDN line interface unit 122, inaccordance with control data such as call requests from the CPU 111inputted via the CPU bus 160.

The ISDN line interface unit 122 is a standard ISDN line interface shownat the regulated S/T point.

The network clock control unit 123 generates a clock which issynchronized to the transfer rate of the ISDN line, which it supplies tothe line switching connection processing unit 121 and the AVmultiplexer/separator unit 124.

The AV multiplexer/separator unit 124 is realized by a specialized LSIcircuit, and separates the AV multiframes sent on the B channel of theISDN line into video code and audio code, which it inputs into the videocodec unit 130 and audio codec unit 140, respectively, via the timesharing bus 162. It also converts audio code from the audio codec unit140 and video code from the video codec unit 130 received via the timesharing bus 162, along with a frame alignment signal (FAS) and abit-rate allocation signal (BAS), into AV multiframes, which it thentransfers to the ISDN line interface unit 122.

The video codec unit 130 is comprised of a D/A convertor 131, an A/Dconvertor 132, a video clock generation unit 133, and video codeccircuit 134. The video codec unit 130 is also connected to the camera151. The video codec unit 130 performs the compression and encoding ofvideo data and the expansion and decoding of video code, based on thevideo sampling signal which is synchronized with the video signal fromthe camera 151. In the following explanation, video informationexpressed as a digital value which has undergone sampling by the videosignal will be referred to as "video data", while compressed and encodedimage data will be referred to as "video code".

The D/A convertor 131 converts the video data expanded and decoded bythe video codec circuit 134 into an analogue signal and sends it to thedisplay control unit 113.

The A/D convertor 132 converts the analogue signal received from thecamera 151 in accordance with the video sampling signal, in doing sogenerating video data.

The video codec circuit 134 has a memory capacity for one second offrame data, and compresses and encodes the video data received from theA/D convertor 132 in accordance with CCITT recommendation H.261. It alsoexpands and decodes the received video code based on the same standard.

The audio codec unit 140 performs the expansion and decoding of audiodata, based on the audio sampling signal which is synchronized with thetransfer speed of the ISDN line which is supplied by the network clockcontrol unit 123. Accordingly, the audio sampling signal is a clocksignal having a frequency of 8 kHz.

The D/A convertor 141 converts the audio data decoded by the audio codeccircuit 143 into an analogue signal, based on the audio sampling signal,and sends it to the speaker 153.

The A/D convertor 142 samples the analogue signal received from themicrophone 151 and converts it into a digital signal in accordance withthe audio sampling signal, in doing so generating audio data.

The audio codec circuit 143 compresses and encodes the audio datareceived from the A/D convertor 142 based on TTC standard JT-G722, G728or suchlike. It also decodes and expands the audio code which isseparated from the received AV multiframes, based on a said samestandard.

In the following explanation, audio information expressed as a series ofdigital values which has undergone sampling by the audio signal will bereferred to as "audio data", while expanded and decoded audio data willbe referred to as "audio code".

The display unit 150, is comprised, for example, of a CRT or an LCD.

The camera 151 is comprised of an image pick-up device such as a CCD(charge couple device).

The microphone 152 converts the inputted sound into an audio signalwhich is an electric signal, which it then outputs to the A/D convertor142.

The speaker 153 reproduces the audio information in the audio signalreceived from the D/A convertor 141.

The CPU interface 160 is connected to the computer bus 114 and thecontrol bus 161, and allows the transfer of control data between the PC110 and the AV unit 170.

The control bus 161 transmits the control data between the CPU interface160 and every component in the AV unit 170.

The time sharing bus 162 is connected to the AV multiplexer/separatorunit 124, the video codec circuit 134 and the audio codec circuit 143,and transfers video code and audio code using a time sharing method.

In the PC-based teleconference terminal 100 constructed as describedabove, AV multiframes received on the B channel of the ISDN line areseparated by the AV multiplexer/separator unit 124, with the video codeand audio code being sent directly to the video codec unit 130 and theaudio codec unit 140, respectively, via the time sharing bus 162,without passing through the PC 110. In the same way, video code from thevideo codec unit 130 and audio code from the audio codec unit 140 aresent directly via the time sharing bus 162, without passing through thecomputer PC 110, to the AV multiplexer/separator unit 124 where they areconverted into multiframes are transmitted on the ISDN line.

However, for the PC-based teleconference terminal 100 under the relatedart, in order to execute teleconferencing, it has been necessary toprovide a general use PC with an AV unit 170 which is separate from thePC 110 and which cannot be directly connected to the computer bus in thePC 110, so that such systems have had the drawbacks of high cost,difficulties regarding reduction in scale, and poor potential forexpansion.

More specifically, these problems are as described below.

1. For the PC-based teleconference terminal 100, a specialized networkcontrol unit 120 is provided in the AV unit 170, so that if the PC 110features its own network control unit capable of ISDN line switching,such as an ISDN board, there will be a duplicity of network controlunits capable of ISDN line switching. Therefore, it can be seen that thehardware resources available in the PC 110 are not being used to theirfull potential.

2. Since the multiframe conversion and separation are executed by an AVmultiplexer/separator unit 124 constructed from a special use LSIcircuit which is provided outside the PC 110, the cost of the systembecomes high, and reduction of the scale of the system becomesproblematic.

3. Since transfer of video code and audio code between the specializednetwork control unit 120, the video codec unit 130 and the audio codecunit 140 is executed without passing through the PC 110, it is notpossible for the PC 110 to control the flow of video code and audiocode. Also, in order for the PC 110 to control the flow of video codeand audio code and to store the video code and audio code on the harddisks disposed in the PC 110, a new specialized network control unit120, and specialized interface software become necessary.

4. Since the specialized network control unit 120 is a specializednetwork control unit for the AV unit 170, then it cannot use theresources included in the PC 110, such as network control software.

It is possible to conceive a system which does not make use of aseparate AV unit 170, wherein a method for the transfer of video codeand audio code is achieved with video and audio codec circuits beingconnected to an AV multiplexer/separator unit composed of software inthe PC 110 via the computer bus 114 of the PC 110. However, under such amethod, the audio sampling signal in the audio codec unit 140 for thegeneration of audio code will not be synchronized to the network clockfor the ISDN line, so that when the generated video code and audio codeare reproduced by the receiver, synchronization slips will occur. Insuch a case, since especially audio data is reproduced for serial bits,then there will be the problems of audio interference and gaps in theaudio signal.

SUMMARY OF THE INVENTION

It is the object of the present invention to provide a teleconferenceterminal which can overcome the problems such as gaps in the audiosignal, and which makes effective use of the hardware and softwareresources in a PC, and so is of low cost and compact size, as well-asallowing for the expansion of its capacity.

The above object can be achieved by a teleconference terminal using apersonal computer for sending and receiving video code, audio code anddata to and from another teleconference terminal, using an ISDN lineswitching method wherein AV multiframes are defined for an H seriesrecommendation for CCITT, comprising: an audio coding unit for encodingand compressing an inputted audio signal in accordance with an audiosampling signal, for generating transmission audio code and outputtingthe transmission audio code to a computer bus, as well as receivingreproduction audio code from the computer bus, and generating areproduction audio signal by decompressing and decoding the reproductionaudio code in accordance with the audio sampling signal; a video codingunit for encoding and compressing an inputted video signal, forgenerating transmission video code and outputting the transmission videocode to the computer bus, as well as receiving reproduction video codefrom the computer bus, and generating a reproduction video signal byexpanding and decoding the reproduction video code; an AV multiframeconversion unit for generating transmission AV multiframes by performingmultiplex conversion of the transmission audio code and the transmissionvideo code inputted from the computer bus according to a CCITTrecommendation, and outputting the generated transmission AV multiframesto the computer bus; a transfer unit for transmitting the transmissionAV multiframes inputted from the computer bus via an ISDN line to theother teleconference terminal, and for receiving reproduction AVmultiframes from the other teleconference terminal according to a CCITTrecommendation via the ISDN line, and outputting the receivedreproduction AV multiframes to the computer bus; an AV separation unitfor receiving the received reproduction AV multiframes from the computerbus and for separating the reproduction video code and the reproductionaudio code from the received reproduction AV multiframes, as well asoutputting the reproduction video code and reproduction audio code tothe computer bus; an internal clock supplying unit for independentlygenerating an internal clock for the personal computer, as well assupplying the audio sampling signal synchronized with the internal clockto the audio coding unit, wherein the audio sampling signal has afrequency of 8 kHz; and an adjustment unit for adjusting surpluses andshortages in the transmission audio code to be included in thetransmission AV multiframes, which occur due to synchronization slipsbetween a network clock of the ISDN line and the internal clock, andsurpluses and shortages in the reproduction audio code separated fromthe reproduction AV multiframes, which occur due to synchronizationslips between the network clock of the ISDN line and the internal clock.

By using the construction described above, data such as audio code,video code, and AV multiframes can be inputted into the PC using thecomputer bus, so that by having the CPU execute programs stored in thePC's memory, a variety of processes such as a synchronization slipcontrol process or the multiplex formation or separation of AVmultiframes can be performed for the data, thereby using the PC'ssoftware resources effectively. Additionally, processes which wereexecuted using LSI in the related art can be achieved by the CPU, sothat the potential for expansion is increased, and reductions in thesize and cost of the teleconference terminal corresponding to the sizeand cost of the specialized LSI can be made.

Also, the teleconference terminal may not comprise an internal clocksupplying unit and an adjustment unit, wherein the transfer unit mayinclude a clock distribution circuit for distributing the network clockof the ISDN line, and a special connector may be formed in each of thetransfer unit and the audio coding unit, with a private clock signalline connected between the special connectors, so that the network clockdistributed by the clock distribution circuit may be supplied via theprivate clock signal line to the audio coding unit.

Also, the count-up pulse generation unit may include: an I/O decoder fordecoding a transfer address of the fixed amount of the reproductionaudio code transferred via the computer bus, and, when the transferaddress is the audio coding unit, for generating a write pulse forwriting data in the reproduction audio code into the audio coding unit,as well as generating a register write pulse; an audio setting registerfor retrieving a number of audio samples written at a start of thereproduction audio code in accordance with the register write pulse andfor storing the number of audio samples; a flip flop which is setaccording to the write pulse for the audio coding unit and reset by aclear signal; a counter for counting up a number of clock pulsesinputted while the flip flop is set, and for being reset to 1! when theflip flop is reset; an agreement circuit for outputting the clear signalto the flip flop when a count value of the counter is equal to thenumber of audio samples stored by the audio setting register; an ANDcircuit for outputting the clock pulses as the count-up signal while theflip flop is set; and a clock oscillator for supplying the clock pulsesto the flip flop, the counter and the AND gate.

By means of the above constructions, the additional benefit of havingthe audio sampling signal synchronized to the network clock can beattained, so that synchronization slips do not occur.

BRIEF DESCRIPTION OF THE DRAWINGS

These and other objects, advantages and features of the invention willbecome apparent from the following description taken in con 3unctionwith the accompanying drawings which illustrate a specific embodiment ofthe invention. In the drawings:

FIG. 1 is a block diagram showing a teleconference terminal constructedaccording to the related art;

FIG. 2 is a block diagram showing the construction of the PC-basedteleconference terminal 200 constructed according to the firstembodiment of the present invention;

FIG. 3 is a figure showing the data format of AV multiframes for CCITT Hseries recommendation on the base ISDN line;

FIG. 4 is a figure showing the data format of the control informationstored by the transmission video descriptor 400, the transmission audiodescriptor 430, and the transmission descriptor 450;

FIG. 5 is a figure showing the data format of the control informationstored by the reception video descriptor 500, the reception audiodescriptor 530, and the reception descriptor 550;

FIG. 6 is a flowchart for the process of the generation and transmissionof AV multiframes by the PC-based teleconference terminal 200 of thepresent embodiment;

FIG. 7 is a flowchart for the process of the reception and separation ofAV multiframes by the PC-based teleconference terminal 200 of thepresent embodiment;

FIG. 8 is a block diagram showing the construction of the PC-basedteleconference terminal 800 according to the second embodiment of thepresent invention;

FIG. 9 is a block diagram showing the construction of the PC-basedteleconference terminal 900 according to the third embodiment of thepresent invention;

FIG. 10 is a block diagram showing the detailed construction of the PLLaudio clock generation unit 941;

FIG. 11 is a block diagram showing the construction of the PC-basedteleconference terminal 1100 according to the fourth embodiment of thepresent invention;

FIG. 12 is a block diagram showing the detailed construction of busclock generation unit 1121 and the audio clock generation unit 1141;

FIG. 13 is a block diagram showing the construction of the PC-basedteleconference terminal 1300 according to the fifth embodiment of thepresent invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS Embodiment 1

FIG. 2 is a block diagram showing the construction of the PC-basedteleconference terminal 200 constructed according to the firstembodiment of the present invention. The PC-based teleconferenceterminal 200 of the present embodiment is principally constructed from aPC 110, a video codec unit 130, a network control unit 220 and an audiocodec unit 240. It should be noted that the video codec unit 130, thenetwork control unit 220 and the audio codec unit 240 are allconstructed on the same board, which is then inserted into the expansionslot of the PC 110. Each of the said units are connected to the PC 110using the computer bus 114 which is the signal line of the expansionslot, and are not connected by any special private lines. The PC 110,the video codec unit 130, and the audio codec unit 240 are additionallyconnected, in the same way as the PC-based teleconference terminal 100,to the display unit 150, the camera 151, the microphone 152 and thespeaker 153. The synchronization slip control software 211 and themultiplexer/separator software 212 are stored in the memory 112. In thisFIG. 2, components which are the same as those shown in FIG. 1 have beengiven the same reference numerals, and their explanation has beenomitted.

The main memory, composed, for example, of DRAM, includes storage areassuch as the transmission video buffer 404, the transmission audio buffer440, the transmission buffer 454, the reception video buffer 504, thereception audio buffer 540 and the reception buffer 555. Also, as a newfunction, the memory 112 stores the synchronization slip controlsoftware 211 and the multiplexer/separator software 212.

The computer bus 114 is the expansion bus for the PC 110, for example,the AT bus used by IBM computers.

The synchronization slip control software 211 is a transmission controlprogram stored by the memory 112. By the CPU 111 executing thesynchronization slip control software 211, the effects on audioreproduction of shortages and surpluses of audio information (calledsynchronization slips), which arise when the audio sampling rate of theaudio codec unit 240 under the present invention cannot be set equal tothe network transmission rate of the ISDN line, can be reduced.

More specifically, when the amount of audio code 441 or audio code 541which is queued in the transmission audio buffer 440 or in the receptionaudio buffer 540 is below a value set as the low mark 433 or the lowmark 533, the CPU 110 judges that such a synchronization slip will occurdue to the shortage of audio data. When there is a shortage of audiodata due to a synchronization slip on transmission, then the CPU 111sends the audio code with the lowest amount of audio code out of all ofthe queued audio code in the transmission audio buffer 440 twice to thetransmission buffer 454. Alternatively, the CPU 111, may calculate theaverage of the audio sample values included in the audio code with alowest amount of sound energy and the audio code queued so as to followthe audio code with a lowest amount of sound energy, and then generateaudio code whose audio code is set at this average value, with thisaudio code then sent following the audio code with a lowest amount ofsound energy to the transmission buffer 454.

Also, when there is a shortage of audio data due to a synchronizationslip on reception, then the CPU 111 sends the audio code with the lowestamount of audio code out of all of the queued audio code in thereception audio buffer 540 twice to the audio codec circuit 143.Alternatively, the CPU 110 may calculate the average of the audio samplevalues included in the audio code with a lowest amount of sound energyand the audio code queued so as to follow after the audio code with alowest amount of sound energy, and then generate audio code whose audiocode is set at this average value, with this audio code then sentfollowing the audio code with a lowest amount of sound energy to theaudio codec circuit 143. In doing so, when there is a synchronizationslip due to a shortage of data, the audio data with the lowest amount ofsound energy will be used for stuffing the gaps in the audio data.

In the same way, when the amount of audio code 441 or audio code 541which is queued in the transmission audio buffer 440 or in the receptionaudio buffer 540 is above a value set as the high mark 432 or the highmark 532, the CPU 111 judges that a synchronization slip will occur dueto a surplus of audio data. When there is a synchronization slip due toa surplus of audio data on transmission, then the CPU 111 ignores theaudio code with the lowest amount of audio energy and, by not sending itto the transmission buffer 54, discards the audio code. Alternatively,the CPU 111, may calculate the average of the audio sample valuesincluded in the audio codes queued so as to precede and follow the audiocode with a lowest amount of sound energy, and then generate audio codewhose audio code is set at this average value, with this generated audiocode then being sent to the transmission buffer 454 in place of theaudio code following after the audio code with the lowest amount ofsound energy, with the audio code with the lowest amount of sound energyand the audio code following after the audio code with the lowest amountof sound energy being discarded. In the same way when there is a surplusof audio data due to a synchronization slip on reception, then the CPU111 ignores the audio code with the lowest amount of audio energy and,by not sending it to the audio codec circuit 143, discards the audiocode. Alternatively, the CPU 111, may calculate the average of the audiosample values included in the audio codes queued so as to precede andafter the audio code with a lowest amount of sound energy, and thengenerate audio code whose audio code is set at this average value, withthis generated audio code then being sent to the audio codec circuit 143in place of the audio code following after the audio code with thelowest amount of sound energy, with audio code with a lowest amount ofsound energy and the audio code following after the audio code with thelowest amount of sound energy being discarded.

The multiplexer/separator software 212 is a transmission control programstored by the memory 112. By the CPU 111 executing themultiplexer/separator software 212, the processes generating AVmultiframes or separating multiframes for this invention can beperformed according to H.221 recommendation for the CCITT, in the sameway as with the AV multiplexer/separator unit 124.

The network control unit 220 includes an ISDN line interface unit 122, aline switching connection processing unit 121, and a network clockcontrol unit 123. It includes also includes, as a new component, a lineswitching transparent data processing unit 221 in place of the AVmultiplexer/separator unit 124.

The line switching transparent data processing unit 221 temporarilystores the AV multiframes inputted into the ISDN line interface unit122, as well as splitting them up, for example into sections of 80bytes, and sending them transparently in unrestricted digital, that isto say, as they are, to the computer bus 114. It also temporarily storesthe AV multiframes inputted into from the CPU 111 via the computer bus114 and sends them as they are to the ISDN line interface unit 122.

The network clock control unit 123 generates the internal clocks for theline switching connection processing unit 121 and the line switchingtransparent data processing unit 221 from the network clock inputtedfrom the ISDN line interface unit 122. The network clock is a clockwhich is synchronized with the transfer rate of the ISDN line, so thatit is a clock signal of frequency 8 kHz, otherwise known as octettiming.

The video codec unit 130 is made up of the D/A convertor 131, the A/Dconvertor 132, the video clock generation unit 133 and the video codeccircuit 134.

The video clock generation unit 133 generates the video sampling signalsynchronized with the video signal input from the camera 151. The videosampling signal is, for example, a sampling signal of frequency 13.5 MHzin accordance with H series recommendation for the CCITT.

The video codec circuit 134 includes a buffer memory (FIFO) for thevideo code sent to and received from the CPU 111.

The audio codec unit 240 is made up of the D/A convertor 141, the A/Dconvertor 142, the audio codec circuit 143 and the audio clockgeneration unit 241.

The audio clock generation unit 241 generates the audio sampling signalthrough self-excited oscillation. This audio sampling signal is a clocksignal of frequency 8 kHz which is independent of the transfer rate ofthe ISDN line.

FIG. 3 is a figure showing the data format of AV multiframes for the Hseries recommendation for the CCITT on the basic rate of the ISDN line(128 kbps). Bit 1 to bit 8 on the first B channel and the second Bchannel are called the subchannels. The network control unit 220 sendsthe audio code A using subchannels 1 and 2 on the first B channel, andsends data D on subchannel 8 on the second B channel in the same way theframe alignment signal (FAS) and the bit-rate allocation signal (BAS)are sent using the subchannel 8 on the first B channel and the second Bchannel, with the rest of the subchannels being used to transmit videocode V.

FIG. 4 is a figure showing the data construction of the controlinformation stored by the transmission video descriptor 400, thetransmission audio descriptor 430, and the transmission descriptor 450.

The transmission video descriptor 400 is a storage area provided in thememory 112 and stores N video buffer descriptors 401.

The N video buffer descriptors 401 are the control information forallowing the CPU 111 to control the N pieces of video code 405 byqueuing in the transmission video buffer 404, and are each made up of avideo buffer pointer 402 and a status 403.

The video buffer pointer 402 shows the address in the transmission videobuffer 404 where the video code 405 corresponding to each video bufferdescriptor 401 is stored.

The status 403 shows whether video code 405 for transmission is presentat the address shown by the video buffer pointer 402.

The transmission video buffer 404 is a storage area provided in thememory 112 as the working storage for the CPU 111 to generate the AVmultiframes using the synchronization slip control software 211 and themultiplexer/separator software 212, and stores N pieces of video code405 for transmission.

The video code 405 is the actual video code of variable length which isencoded by the video codec circuit 134.

The transmission audio descriptor 430 is a storage area provided in thememory 112 and stores the audio synchronization control information 431and M audio buffer descriptors 435.

The audio synchronization control information 431 is the controlinformation for compensating for the effects on audio reproductioncaused by synchronization slips, and is made up of a high mark 432, alow mark 433 and an insertion segment pointer 434.

The high mark 432 shows the predetermined upper limit for use of thetransmission audio buffer 440, which is for overseeing the actual usedamount of the transmission audio buffer 440 during transmission. Forexample, when the transmission audio buffer 440 is capable of storing 8pieces of audio code 441, and an upper limit for the storage of audiocode 441 is set as 5 pieces, then the high mark 432 shows 5!.

The low mark 433 shows the predetermined lower limit for use of thetransmission audio buffer 440, which is for overseeing the actual usedamount of the transmission audio buffer 440 during transmission. Forexample, when the transmission audio buffer 440 is capable of storing 8pieces of audio code 441, and a lower limit for the storage of audiocode 441 is set as 1 piece, then the low mark 433 shows 1!.

The insertion segment pointer 434 shows the segment number of the audiocode segment 442 for which insertion is to be performed, when the CPU111 detects a shortage of audio data due to a synchronization slip andso determines insertion into an audio code segment 442 is to beperformed. Audio code segments 442 whose segment numbers are set as theinsertion segment pointer 434 are sent twice to the to the transmissionbuffer 454. Alternatively, the CPU 111, may calculate the average of theaudio sample values included in the audio code segment 442 whose segmentnumber is stored in the insertion segment pointer 434 and the audio codequeued so as to follow after the audio code whose segment number isstored in the insertion segment pointer 434, and then generate a newaudio code segment 442 whose audio sample is set at this average value,with this audio code segment 442 then being sent following the audiocode segment 442 whose segment number is stored in the insertion segmentpointer 434 to the transmission buffer 454.

The M audio buffer descriptors 435 are the control information forallowing the CPU 111 to control the M pieces of audio code 441 each madeup of S audio code segments 442 in the transmission audio buffer 440,and are each made up of S segment descriptors 436.

The segment descriptors 436 are the control information obtained bydividing the audio code 441 in segments for allowing management in termsof units of one audio code segment 442, and are each made up of an audiobuffer pointer 437, a status 438 and a priority 439.

The audio buffer pointer 437 shows the address of the segment in thetransmission audio buffer 440 which stores the audio code segment 442corresponding to the segment descriptor 436 in question.

The status 438 stores for the segment indicated by the audio bufferpointer 437 one of the following three possible statuses: "Correspondingaudio code segment 442 present", "No corresponding audio code segment442 present", or "Corresponding audio code segment 442 present, butinvalid".

The priority 439 shows the sum of the squares of the audio sample valuesincluding in the corresponding audio code segment 442. This shows theamount of sound energy present in the corresponding audio code segment442.

The transmission audio buffer 440, in the same way as the transmissionvideo buffer 404, is a storage area provided in the memory 112 as theworking storage for the CPU 111 to generate the AV multiframes, andstores the audio code 441 in units of one segment.

The audio code 441 is the encoded actual audio code, and is made up of Saudio code segments 442. More specifically, the audio code 441, if, forexample, made up of 80 bytes of audio code, will be divided into 8 audiocode segments 442, each being of 10 bytes long.

The audio code segments 442 are the segments of audio code created bydividing the audio code 441.

The transmission descriptor 450 is a storage area provided in the memory112 and stores K transmission buffer descriptors 451.

The transmission buffer descriptors 451 are the control information forallowing the CPU 111 to manage of K pieces of transmission code 455 byqueuing them, and are each made up of a transmission buffer pointer 452and a status 453.

The transmission buffer pointer 452 shows an address in the transmissionbuffer 454 at which the corresponding transmission code 455 is stored.

The status 453 shows whether there is corresponding transmission code455 stored at the address in the transmission buffer 454 shown by thetransmission buffer pointer 452.

The transmission buffer 454 stores the AV multiframes for transmissionmade up of the K pieces of transmission code 455 which are created bythe multiplex conversion of the audio code, the video code and data. Thestored K pieces of transmission code 455 are queued for transmission andthen sent to the network control unit 220.

FIG. 5 is a figure showing the data construction of the controlinformation stored by the reception video descriptor 500, the receptionaudio descriptor 530, and the reception descriptor 550.

The reception video descriptor 500 is a storage area provided in thememory 112 and stores N video buffer descriptors 501.

The N video buffer descriptors 501 are the control information forallowing the queuing in the reception video buffer 504 of the video code505 separated from the received AV multiframes, and are each made up ofa video buffer pointer 502 and a status 503.

The video buffer pointer 502 shows the address in the reception videobuffer 504 where the video code 505 corresponding to each video bufferdescriptor 501 is stored.

The status 503 shows whether received video code 505 is present at theaddress shown by the video buffer pointer 502.

The reception video buffer 504 stores N pieces of received video code505 separated from the received AV multiframes.

The video code 505 is the actual video code for reproduction which isseparated from the received AV multiframes by means of the CPU 111executing the multiplexer/separator software 212.

The reception audio descriptor 530 stores the audio synchronization,control information 531 and M audio buffer descriptors 535.

The audio synchronization control information 531 is the controlinformation for compensating for the effects on audio reproductioncaused by synchronization slips, and is made up of a high mark 532, alow mark 533 and an insertion segment pointer 534.

The high mark 532 shows the predetermined upper limit for use of thereception audio buffer 540, which is for overseeing the actual usedamount of the reception audio buffer 540 during reception. For example,when the reception audio buffer 540 is capable of storing 8 pieces ofaudio code 541, and an upper limit for the storage of audio code 541 isset as 5 pieces, then the high mark 532 shows 5!.

The low mark 533 shows the predetermined lower limit for use of thereception audio buffer 540, which is for overseeing the actual usedamount of the reception audio buffer 540 during reception. For example,when the reception audio buffer 540 is capable of storing 8 pieces ofaudio code 541, and a lower limit for the storage of audio information541 is set as 1 piece, then the low mark 533 shows 1!.

The insertion segment pointer 534 shows the segment number of the audiocode segment 542 for which insertion is to be performed, when the CPU111 detects a shortage of audio data due to a synchronization slip andso determines insertion into an audio code segment 542 is to beperformed. Audio code segments 542 whose segment numbers are set as theinsertion segment pointer 534 are sent twice to the to the audio codeccircuit 143. Alternatively, the CPU 111, may calculate the average ofthe audio sample values included in the audio code segment 542 whosesegment number is stored in the insertion segment pointer 534 and theaudio code segment 542 queued so as to follow after the audio, codewhose segment number is stored in the insertion segment pointer 534, andthen generate a new audio code segment 542 whose audio sample is set atthis average value, with this audio code segment 542 then being sentfollowing the audio code segment 542 whose segment number is stored inthe insertion segment pointer 534 to the audio codec circuit 143.

The M audio buffer descriptors 535 are the control information forallowing the CPU 111 to control the M pieces of audio code 541 each madeup of S audio code segments 542 in the reception audio buffer 540, andare each made up of S segment descriptors 536.

The segment descriptors 536 are the control information obtained bydividing the audio code 541 in segments for allowing management in termsof units of one audio code segment 542, and are each made up of an audiobuffer pointer 537, a status 538 and a priority 539.

The audio buffer pointer 537 shows the address of the segment in thereception audio buffer 540 which stores the audio code segment 542corresponding to the segment descriptor 536 in question.

The status 538 stores for the segment in the reception audio buffer 540indicated by the audio buffer pointer 537 one of the following threepossible statuses: "Corresponding audio code segment 542 present", "Nocorresponding audio code segment 542 present", or "Corresponding audiocode segment 542 present, but invalid".

The priority 539 shows the sum of the squares of the audio sample valuesincluding in the corresponding audio code segment 542. This shows theamount of sound energy present in the corresponding audio code segment542.

The reception audio buffer 540 stores the M pieces of audio code 541separated from the received AV multiframes.

The audio code 541 is the actual audio code separated from the receivedAV multiframes, and is made up of S audio code segments 542. Morespecifically, the audio code 541, if for example made up of 80 bytes ofaudio code, will be divided into 8 audio code segments 542, each beingof 10 bytes long.

The audio code segments 542 are the segments of audio code created bydividing the audio code 541.

The reception descriptor 550 is a storage area provided in the memory112 and stores the frame alignment pointer 551 and K reception bufferdescriptors 552.

The frame alignment pointer 551 shows the address in the receptionbuffer 555 shown in the frame alignment signal FAS.

The reception buffer descriptors 552 are the control information forallowing the management by queuing of K pieces of reception code 556inputted as AV multiframes from the network control unit 220, and areeach made up of a reception buffer pointer 553 and a status 554.

The reception buffer pointer 553 shows an address in the receptionbuffer 555 at which the corresponding reception code 556 is stored.

The status 554 shows whether there is corresponding reception code 556stored at the address in the reception buffer 555 shown by the receptionbuffer pointer 553.

The reception buffer 555 stores the AV multiframes inputted from thenetwork control unit 220 made up of the K pieces of reception code 556.

The reception code 556 is the AV multiframes received from the networkcontrol unit 220.

FIG. 6 is a flowchart for the process of the generation and transmissionof AV multiframes by the PC-based teleconference terminal 200 of thepresent embodiment. The process described below is performed by the CPU111 executing the synchronization slip control software 211 and themultiplexer/separator software 212.

The CPU 111 receives an input ISDN number, indicating a teleconferenceterminal which is to be accessed, from a keyboard or suchlike (notillustrated) which is connected to the computer bus 114, and, by usingthe D channel on the ISDN line, establishes a connection with theaccessed terminal via the line switching connection processing unit 121.Having established the connection, the PC 110 executes the processesdescribed below using one or a number of B channels, with video code,audio code, and data being transmitted and received in the AV multiframeformat shown in FIG. 3. The transmitting and receiving of video code,audio code, and data between the CPU 111 and the ISDN line interfaceunit 122 is executed transparently via the line switching transparentdata processing unit 221 and the computer bus 114.

In the video codec unit 130, the video signal from the camera 151 is ADconverted by the A/D convertor 132, with the AD converted video signalthen being compressed and encoded by the video codec circuit 134,thereby generating the video code 405. Having waited until space forstoring the video code 405 in the transmission video buffer 404 isavailable (S601), the CPU 111 has the video code 405 sent via thecomputer bus 114 to the memory 112 where it is queued in thetransmission video buffer 404 as shown in FIG. 4 (S602).

In the same way, in the audio codec unit 240 the audio signal from themicrophone 152 is AD converted by the A/D convertor 142, with the ADconverted audio signal then being compressed and encoded by the audiocodec circuit 143, thereby generating the audio code segments 442.Having waited until space for storing the audio code segments 442 in thetransmission audio buffer 440 is available (S603), the CPU ill has theaudio code segments 442 sent via the computer bus 114 to the memory 112where they are queued in the transmission audio buffer 440 as shown inFIG. 4 (S604). In doing so, when a fixed amount of video code 405 andaudio code 441 are queued in the transmission video buffer 404 and thetransmission audio buffer 440 for the first time, the process describedbelow is executed.

The CPU 111, in order to compensate for any difference between thetransfer rate of the ISDN line and the transfer rate from the audiocodec unit 240, executes the process described below by executing thesynchronization slip control software 211 for the audio code 441.

The CPU 111 refers to the low mark 433 stored in the transmission audiodescriptor 430 and if the amount of audio code 441 queued in thetransmission audio buffer 440 is less than the low mark 433 which is thelower limit for use of the transmission audio buffer 440 (S605), then itinvestigates whether there is a status 438 which is "Audio code segment442 present in transmission audio buffer 440, but invalid" (S606). Ifthere is such a status 438, then it is changed to "Audio code segment442 present in transmission audio buffer 440", and the process advancesto S615 (S607).

If, in S606, there is no status 438 which is "Audio code segment 442present in transmission audio buffer 440, but invalid", then the CPU 111calculates the sound energy in each audio code segment 442 of the queuedaudio code 441, and, in addition to setting the priority 439 in thecorresponding segment descriptor 436 (S608), sets the segment number ofthe audio code segment 442 with the lowest priority 439, out of all ofthe audio code segments 442 queued in the transmission audio buffer 440,in the insertion segment pointer 434, before moving on to S615 (S609).

In S605, if the amount of audio code 441 queued in the transmissionaudio buffer 440 is above the low mark 433, then the CPU 111 refers tothe high mark 432, and investigates whether the amount of audio code 441queued in the transmission audio buffer 440 is more than the high mark432 which is the upper limit for use of the transmission audio buffer440 (S610). If it is more than the high mark 432, then the CPU 111investigates whether there is a segment number set in the insertionsegment pointer 434 (S611), clearing such numbers if present, beforemoving on to S615 (S612).

If in S611 there is no segment number set in the insertion segmentpointer 434, then the CPU 111 calculates the sound energy in each audiocode segment 442 of the queued audio code 441, and, in addition tosetting the priority 439 in the corresponding segment descriptor 436(S613), sets the status 438 in the segment descriptor 436 correspondingto the audio code segment 442 with the lowest priority 439, out of allof the audio code segments 442 queued in the transmission audio buffer440, as "Corresponding audio code segment 442 present, but invalid"(S614).

The CPU 111 then waits until the video code 405 and the audio code 441have been queued in the transmission video buffer 404 and in thetransmission audio buffer 440 in the memory 112 (S615), before waitinguntil transmission to the network control unit 220 becomes possible(S616), and, when transmission to the network control unit 220 becomespossible, it transmits the video code 405 from the transmission videobuffer 404 to the video area of the transmission buffer 454 (S617).

The CPU 111 then transfers the audio code segments 442 in thetransmission audio buffer 440 to the audio area of the AV multiframes,shown in FIG. 3, in the transmission buffer 454, and generates the AVmultiframes, at the same time queuing them temporarily in thetransmission buffer 454. At this point, the CPU 111 arranges the audiocode 441 so as to be delayed, for example, by about 10 msec, withrespect to the video code 405 in the AV multiframes as a lip-synchprocess (the synchronization of audio and video to be reproduced). Thisis because the coding rate of the video codec circuit 134 is slower thanthe coding rate of the audio codec circuit 143, for example, by about 10msec per AV multiframe.

In the same way, when the segment number of the audio code segment 442to be transmitted is written in the insertion segment pointer 434, theCPU 111 transfers such an audio code segment 442 2 times consecutivelyto the transmission buffer 454, with the insertion segment pointer 434in question being cleared once the second transfer of the audio codesegment 442 has been completed. Alternatively, the CPU 111, maycalculate the average of the audio sample values included in the audiocode segment 442 whose segment number is stored in the insertion segmentpointer 434 and the audio code queued so as to follow after the audiocode whose segment number is stored in the insertion segment pointer434, and then generate a new audio code segment 442 whose audio sampleis set at this average value, with this audio code segment 442 thenbeing sent following the audio code segment 442 whose segment number isstored in the insertion segment pointer 434 to the transmission buffer454. On such an occasion, then once the transmission of the new audiocode segment 442 is complete, then the corresponding insertion segmentpointer 434 is cleared.

Additionally, if at this point the status 438 of the audio code segment442 to be transferred is "Audio code segment 442 present in transmissionaudio buffer 440, but invalid", then such an audio code segment 442 isnot transferred to the transmission buffer 454 and is discarded.Alternatively, the CPU 111, may calculate the average of the audiosample values included in the audio code segments 442 which precede andfollow the audio code segment 442 whose status 438 is "Audio codesegment 442 present in transmission audio buffer 440, but invalid", andthen generate a new audio code segment 442 whose audio sample is set atthis average value, with this generated audio code segment 442 thenbeing transferred to the transmission buffer 454 in place of the audiocode segment 442 queued so as to follow the audio code segment 442 whosestatus 438 is "Audio code segment 442 present in transmission audiobuffer 440, but invalid", and the audio code segment 442 whose status438 is "Audio code segment 442 present in transmission audio buffer 440,but invalid" and the following audio code segment 442 being discarded.

The CPU 111 transfers or discards each of the audio code segments 442 tothe audio code segment 442 and updates the corresponding segmentdescriptors 436 (S618).

The CPU 111 then forms the AV multiframes in the transmission buffer 454and, having queued the transmission code 455, transfers it consecutivelyto the network control unit 220 (S619), with the network control unit220 sending these AV multiframes out on one B channel or a number of Bchannels on the ISDN line.

The process shown as S601 through S619 is then repeated for thegeneration of each piece of video code 405 and each audio code segment442 by the video codec circuit 134 and the audio codec circuit 143.

FIG. 7 is a flowchart for the process of the reception and separation ofAV multiframes by the PC-based teleconference terminal 200 of thepresent embodiment. The following process is performed, in the same wayas with transmission, by having the CPU 111 execute themultiplexer/separator software 212 and the synchronization slip controlsoftware 211.

The CPU 111 first investigates whether there is an empty storage area inthe reception buffer 555 (S701), and if there is, transfers the receivedAV multiframes via the line switching transparent data processing unit221 to the memory 112. The received AV multiframes are then queued intheir multiframe state in the reception buffer 555 shown in FIG. 5 inblock units of a fixed length (S702).

By executing the multiplexer/separator software 212, the CPU 111searches for the frame alignment signal (FAS) in the content of thereception buffer 555, and taking the frame alignment, stores it as theframe alignment pointer 551 in the reception descriptor 550, as well asseparating the audio code 541, the video code 505, the data, and thebit-rate allocation signal (BAS) (S703).

The CPU 111 investigates whether there is an empty storage area in thereception video buffer 504 and in the reception audio buffer 540 (S704),and if there is, executes a lip-synch process, more specifically thedelaying of the separated audio code 541 of about 10 msec with regard tothe video code 505, and queues the video code 505 in the reception videobuffer 504 and the audio code 541 in the reception audio buffer 540,respectively (S705).

It should be noted that the CPU 111 performs the followingsynchronization slip process in order to compensate for any differencein the reception rate from the ISDN line and the transfer rate to theaudio codec unit 240 by executing the synchronization slip controlsoftware 211. The synchronization slip process S706 through S715described below is executed according to the same process as for theprocess executed during the generation of AV multiframes S605 to S614.

The CPU 111 refers to the low mark 533 stored in the reception audiodescriptor 540 and if the amount of audio code 541 queued in thereception audio buffer 540 is less than the low mark 533 (S706), then itinvestigates whether there is a status 538 which is "Audio code segment542 present in reception audio buffer 540, but invalid" (S707). If thereis such a status 538, then it is changed to "Audio code segment 542present in reception audio buffer 540", and the process advances to S716(S708).

If, in S707, there is no status 538 which is "Audio code segment 542present in reception audio buffer 540, but invalid", then the CPU 111calculates the sound energy in each audio code segment 542 queued in thereception audio buffer 540 in the same way as during transmission, and,in addition to setting the priority 539 in the corresponding segmentdescriptor 536 (S709), sets the segment number of the audio code segment542 with the lowest priority 539, out of all of the audio code segments542 queued in the transmission audio buffer 540, in the insertionsegment pointer 534 in the reception audio descriptor 530, before movingon to S716 (S710).

In S706, if the Amount of audio code 541 queued in the reception audiobuffer 540 is above the low mark 533 in the reception audio descriptor530, then the CPU 111 refers to the high mark 532 in the reception audiodescriptor 530, and investigates whether the amount of audio code 541queued in the reception audio buffer 540 is more than the high mark 532(S711). If it is more than the high mark 532, then the CPU 111investigates whether there is a segment number set in the insertionsegment pointer 534 (S712), clearing such numbers if present, beforemoving on to S716 (S713).

If in S712 there is no segment number set in the insertion segmentpointer 534, then the CPU 111 calculates the sound energy in each queuedaudio code segment 542, and, in addition to setting the priority 539(S714), sets the status 538 in the segment descriptor 536 correspondingto the audio code segment 542 with the lowest priority 539, out of allof the audio code segments 542 queued in the reception audio buffer 540,as "Corresponding audio code segment 542 present, but invalid" (S715).

The CPU 111 then waits until the audio codec unit 240 is able to receivethe audio code segments 542 (S716) and transfers successively the audiocode segments 542 to the audio codec unit 240, based on the contents ofthe reception audio descriptor 530 by executing themultiplexer/separator software 212. When transferring audio codesegments 542 whose segment number is stored in the insertion segmentpointer 534, the CPU 111 sends such audio code segments 542consecutively twice. Alternatively, the CPU 111, may calculate theaverage of the audio sample values included in the audio code segment542 whose segment number is stored in the insertion segment pointer 534and the audio code segment 542 queued so as to follow after the audiocode segment 542 whose segment number is stored in the insertion segmentpointer 534, and then generate a new audio code segment 542 whose audiosample is set at this average value, with this new audio code segment542 then being sent following the audio code segment 542 whose segmentnumber is stored in the insertion segment pointer 542 to the audio codeccircuit 143.

In the same way, when the status 538 of the audio code segment 542 is"Corresponding audio code segment 542 present, but invalid", then theCPU 111 does not transfer the audio code segment 542 in question.Alternatively, the CPU 111, may calculate the average of the audiosample values included in the audio code segments 542 which precede andfollow the audio code segment 542 whose status 538 is "Audio codesegment 542 present in transmission audio buffer 540, but invalid" andthe audio code segment 542 queued so as to follow after such an audiocode segment 542, and then generate a new audio code segment 542 whoseaudio sample is set at this average value, with this generated audiocode segment 542 then being sent to the audio codec circuit 143 in placeof the audio code segment 542 queued so as to follow the audio codesegment 542 for which the status 539 is "Audio code segment 542 presentin transmission audio buffer 540, but invalid", and the audio codesegment 542 whose status 538 is "Audio code segment 542 present intransmission audio buffer 540, but invalid" and the following audio codesegment 542 then being discarded. In the same way, the CPU 111, ontransferring each audio code segment 542 in the reception audio buffer540, updates the corresponding segment descriptor 536 and the insertionsegment pointer 534 (S717).

The CPU 111 then transfers the video code 505 queued in the receptionvideo buffer 504 to the video codec unit 130 (S718).

In the same way, if in S710 or in S704 there is no empty storage area inany of the buffers out of the reception buffer 555, the reception videobuffer 504 and the reception audio buffer 540, then the CPU 111 judgesthat the video codec circuit 134 or the audio codec circuit 143 isstopped, and, in addition to displaying an error message on the displayunit 150, notifies the other terminals with whom transmission is beingperformed of the occurrence of the error before terminating itsprocessing (S719).

After the above process has been performed, in the video codec unit 130the video code 505 is decoded, DA converted by the D/A convertor 131,and then displayed by the display unit 150 having been superimposed overthe computer screen by the display control unit 113. In the audio codecunit 240, the audio code 541 is decoded, DA converted by the D/Aconvertor 141, and then reproduced by the speaker 153. The process formS701 through S719 is then repeated for the next reception of AVmultiframes by the network control unit 220.

In the way described above, by means of the present embodiment, sincethe PC 110, the video codec unit 130, the network control unit 220 andthe audio codec unit 240 can transfer data including audio code andvideo code between themselves via the computer bus 114, then thePC-based teleconference terminal 200 can make effective use of thehardware and software resources in the PC 110, and the processing forwhich the PC-based teleconference terminal 100 needed a specialized LSIcircuit can be executed by the CPU 111.

That is to say, for the PC-based teleconference terminal 200, themultiplex conversion and separation of video code and audio code can beperformed by the CPU 111 by executing the multiplexer/separator software212. The effects on audio reproduction caused by synchronization slipscan also be prevented by the CPU 111 executing the synchronization slipcontrol software 211. Also, since for the PC-based teleconferenceterminal 200, the PC 110, the video codec unit 130, the network controlunit 220 and the audio codec unit 240 can transfer data including audiocode and video code between themselves via the computer bus 114, thenthe CPU 111 in this embodiment can further partly take over theprocessing such as the generation and decoding of video code and thegeneration and decoding of audio code executed by the video codeccircuit 134 and the audio codec circuit 143. In addition, it is possiblefor the CPU 111 to execute a variety of processes on the video and audiosignals, such as executing audio recognition and displaying the receivedaudio data as subtitles for the hard of hearing.

It should be noted that in this embodiment, the priority 439 has beendescribed as being the sum of the squares of the all of audio samplesincluded in the corresponding audio code segments 442, but this priority439 need not be calculated in such a way. For example, when the audiocodec circuit 143 is using PCM (pulse code modulation) for the G.711recommendation on the CCITT, it is possible for the priority to becalculated as the sum of the squares of the values of the audio samples,or as the average of the squares of the audio sample values. When theaudio codec circuit 143 is using ADPCM (adapted differential pulse codemodulation) for the G.726 recommendation on the CCITT, it is alsopossible for the calculation to be for the sum of the differentialvalues under ADPCM of all of the corresponding audio code segments 442,or for the sum of the squares of the differential values under ADPCM. Inthe same way, the priorities 539 for the audio code segments 542 may becalculated in the same way.

Embodiment 2

FIG. 8 is a block diagram showing the construction of the PC-basedteleconference terminal 800 according to the second embodiment of thepresent invention. The PC-based teleconference terminal 800 of thepresent embodiment is principally constructed of the same components asthe PC-based teleconference terminal 200 shown in FIG. 2, so that suchcomponents have been given the same reference numerals, and theirexplanation has been omitted.

The PC-based teleconference terminal 800 is constructed from a PC 110, avideo codec unit 130, a network control unit 820 and an audio codec unit840. The PC 110, the video codec unit 130, and the audio codec unit 840are all connected, in the same way as the PC-based teleconferenceterminal 200, to the display unit 150, the camera 151, the microphone152 and the speaker 153. In this embodiment, the video codec unit 130,the network control unit 820 and the audio codec unit 840 are allconstructed on the same board, which is then inserted into the expansionslot of the PC 110.

The network control unit 820 includes a line switching connectionprocessing unit 121, an ISDN line interface unit 122, a network clockcontrol unit 823 and a line switching transparent data processing unit221. The network clock control unit 823 additionally includes a samplingsignal generation circuit 824. The audio codec unit 840 includes a D/Aconvertor 141, an A/D convertor 142, and an audio codec circuit 143. Thememory 112 stores the multiplexer/separator software 212.

The differences between the present embodiment and the PC-basedteleconference terminal 200 of the first embodiment are the networkcontrol unit 820 and the audio codec unit 840. Since there is no wiringfor transferring the clock signal and no connector or suchlike forconnecting such a clock signal line on a ready-made expansion board, thenetwork clock control unit 823 and the audio codec unit 840 areconnected to each other by a connector in the form of the private signalline 801.

The sampling signal generation circuit 824 provides the ISDN networkclock as the audio sampling signal to the D/A convertor 141, the A/Dconvertor 142, and the audio codec circuit 143 via the private signalline 801.

The audio code generated by the audio codec unit 840 is temporarilyentered into the memory 112 via the computer bus 114, and is queued inthe transmission audio buffer. In the same way transmission video codegenerated by the video codec unit 130 is queued in the transmissionvideo buffer. AV multiframes are then composed in the transmissionbuffer from the video code queued in the transmission video buffer andthe audio code queued in the transmission audio buffer, and queued,before being sent successively to the line switching transparent dataprocessing unit 221. In this embodiment, however, since the audiosampling signal is synchronized to the ISDN network clock,synchronization slips do not occur. Therefore the transmission processdiffers to that of the first embodiment in that the synchronization slipcontrol process shown as S605 to S614 in FIG. 6 is not performed.

In the same way as in the first embodiment, the AV multiframes receivedby the ISDN line interface unit 122 are queued in the reception bufferprepared in the memory 112 via the computer bus 114. Alignment is takenfrom the frame alignment signal (FAS) and the AV multiframes areseparated into video code, audio code, data and the bit-rate allocationsignal (BAS) with the received video code being queued in the receptionvideo buffer and the received audio code being queued in the receptionaudio buffer. The video code queued in the reception video buffer isthen sent successively to the video codec circuit 134, and having beendecoded by the video codec circuit 134, is DA converted by the D/Aconvertor 131 and is displayed by the display control unit 113 on thedisplay unit 150. The audio code queued in the reception audio buffer isthen sent successively to the audio codec circuit 143, and, having beendecoded by the audio codec circuit 143, is DA converted by the D/Aconvertor 141 and is then reproduced by the speaker 153. It should benoted that in this embodiment synchronization slips do not occur, sothat in the same way as the transmission process, the synchronizationslip control process of S706 through S715 of the first embodiment shownin FIG. 7 is not executed.

In the way described above, the present embodiment has the additionalbenefit of the avoidance of synchronization slips by having the networkclock inputted as the audio sampling signal from the network clockcontrol unit 823 via the private signal line.

Embodiment 3

FIG. 9 is a block diagram showing the construction of the PC-basedteleconference terminal 900 according to the third embodiment of thepresent invention. The PC-based teleconference terminal 900 of thepresent embodiment is principally constructed of the same components asPC-based teleconference terminal 100, the PC-based teleconferenceterminal 200 and the PC-based teleconference terminal 800, so that suchcomponents have been given the same reference numerals, and theirexplanation has been omitted.

The PC-based teleconference terminal 900 is constructed from a PC 110, avideo codec unit 130, a network control unit 220 and an audio codec unit940. The PC 110, the video codec unit 130, and the audio codec unit 940are all connected, in the same way as the PC-based teleconferenceterminal 200, to the display unit 150, the camera 151, the microphone152 and the speaker 153. The memory 112 again stores themultiplexer/separator software 212.

The audio codec unit 940 includes a D/A convertor 141, an A/D convertor142, an audio codec circuit 143 and a PLL audio clock generation unit941.

FIG. 10 is a block diagram showing the detailed construction of the PLLaudio clock generation unit 941.

The PLL audio clock generation unit 941 is made up of an I/O decoder1001, an updown calculator 1008, an integrator 1009, a variable crystaloscillator 1010 and a write data conversion circuit 1011. The write dataconversion circuit 1011 is further comprised of an audio settingregister 1002, an agreement circuit 1003, a counter 1004, a JK-FFcircuit (flip-flop circuit) 1005, an AND gate 1006 and a clocktransmitter 1007.

The I/O decoder 1001 decodes the device address which is the transferaddress of the data input from the computer bus 114, and, if the inputdata is reception audio code to be written into the audio codec circuit143, generates the register write pulse for the audio setting register1002, as well as writing the audio pulse number included in a receptionaudio code at the start of said reception audio code into the audiosetting register 1002. In addition, the I/O decoder 1001 generates thewrite pulse for the audio codec circuit 143, and writes said receptionaudio code into the audio codec circuit 143.

The write data conversion circuit 1011 generates a same number of pulsesas the number of audio samples written into the audio setting register1002.

The audio setting register 1002 stores the number of audio samples inthe audio code written into the audio codec circuit 143 in one writeoperation.

The agreement circuit 1003 generates a clear signal for the JK-FFcircuit 1005 when the number of audio samples stored by the audiosetting register 1002 is equal to the count value of the counter 1004.

The counter 1004 is set to 1! when cleared and, while the output Q ofthe JK-FF circuit 1005 is 1!, that is to say, while the clear signal is0!, counts the number of clock pulses from the clock transmitter 1007.

The JK-FF circuit 1005 has its K terminal earthed, and is set accordingto the write pulse for the audio codec circuit 143, and reset accordingto the clear signal from the agreement circuit 1003. It should be notedthat the JK-FF circuit 1005 need not be composed of an JK-FF circuit,and may be composed of, for example, an RS-FF circuit, so long as it isset according to the write pulse for the audio codec circuit 143, andreset according to the clear signal from the agreement circuit 1003.

The AND gate 1006 outputs the up signal when there is a logical ANDbetween the Q output from the JK-FF circuit 1005 and the clock pulsefrom the clock transmitter 1007.

The clock transmitter 1007 generates a clock signal of around 10 MHz andoutputs this generated clock pulse to the JK-FF circuit 1005, the ANDgate 1006 and the counter 1004.

The updown calculator 1008 outputs the difference between the up signalfrom the AND gate 1006 and the down signal which is the audio samplingsignal from the variable crystal oscillator 1010 to the integrator 1009.

The integrator 1009 is constructed of a low-pass filter or suchlike, andsmoothes the output of the updown calculator 1008, before outputting tothe variable crystal oscillator 1010.

The variable crystal oscillator 1010 is one kind of voltage controloscillator (VCO) which can change its frequency in accordance with theoutput of the integrator 1009, and, in addition to outputting an audiosampling signal of a frequency of about 8 MHz to the updown calculator,outputs to the D/A convertor 141 and the A/D convertor 142.

In the manner described above, the updown calculator 1008, theintegrator 1009, and the variable crystal oscillator 1010 form a phaselock loop (PLL). By doing so, the frequency of the audio sampling signalcan be set equal to the average frequency of the number of audio sampleswritten into the audio codec circuit 143, and, as a result, set equal tothe network clock.

As described above, by means of the present embodiment, in addition tothe effect of the second embodiment, an audio sampling signalsynchronized to the network clock by the PLL audio clock generation unit941 can be generated, so that synchronization slips can be avoided, butunlike the second embodiment, without providing a new terminal forconnecting the private signal line 801, and without executing acomplicated process for operating the private signal line 801, on anexpansion board or a PC card.

Embodiment 4

FIG. 11 is a block diagram showing the construction of the PC-basedteleconference terminal 1100 according to the fourth embodiment of thepresent invention. The PC-based teleconference terminal 1100 of thepresent embodiment is principally constructed of the same components asPC-based teleconference terminal 100, and the PC-based teleconferenceterminal 200, so that such components have been given the same referencenumerals, and their explanation has been omitted.

The PC-based teleconference terminal 1100 is constructed from a PC 110,a video codec unit 130, a network control unit 1120 and an audio codecunit 1140. The PC 110, the video codec unit 130, and the audio codecunit 1140 are all connected, in the same way as the PC-basedteleconference terminal 200, to the display unit 150, the camera 151,the microphone 152 and the speaker 153. The memory 112 again stores themultiplexer/separator software 212.

The network control unit 1120 is made up of a line switching connectionprocessing unit 121, an ISDN line interface unit 122, a network clockcontrol unit 123, a line switching transparent data processing unit 221,and a bus clock generation unit 1121. The audio codec unit 1140 is madeup of a D/A convertor 141, an A/D convertor 142, an audio codec circuit143 and an audio clock generation unit 1141. The computer bus 114 ismade up of a control signal line 1151 and a data bus 1152.

The bus clock generation unit 1121 multiplies the network clock from thenetwork clock control unit 123, and so generates the bus clock for thecomputer bus 114.

The control signal line 1151 carries the control signals, such as thebus clock.

The data bus 1152 carries the audio code, the video code, and otherkinds of data.

The audio clock generation unit 1141 divides the bus clock of thecomputer bus 114 and so generates the audio sampling signal.

FIG. 12 is a block diagram showing the detailed construction of busclock generation unit 1121 and the audio clock generation unit 1141.

The bus clock generation unit 1121 is made up of a phase comparator1221, a variable crystal oscillator 1222, and a frequency divider 1223.

The phase comparator 1221 outputs the phase difference between the 8 kHznetwork clock supplied by the network clock control unit 123 and theoutput of the frequency divider 1223.

The variable crystal oscillator 1222 generates a clock pulse of afrequency of about 10 MHz, and, in addition to outputting this clockpulse to the frequency divider 1223, outputs it as the bus clock to thecontrol signal line 1151.

The frequency divider 1223 divides the frequency of the clock pulse fromthe variable crystal oscillator 1222 by multiplying it by a value 1/N,for example 8/10000.

Since the phase comparator 1221, the variable crystal oscillator 1222,and the frequency divider 1223 form a PLL circuit, the bus clockgeneration unit 1121 can generate a bus clock synchronized with thenetwork clock.

The audio clock generation unit 1141 is made up of a phase comparator1241, a variable crystal oscillator 1242 and a frequency divider 1243.

The phase comparator 1241 outputs the phase difference between the busclock inputted from the control signal line 1151 and the output of thefrequency divider 1243.

The variable crystal oscillator 1242 generates a clock pulse of afrequency of about 8 kHz, and, in addition to outputting this clockpulse to the frequency divider 1243, outputs it as the audio samplingsignal to the D/A convertor 141 and the A/D convertor 142.

The frequency divider 1243 multiplies the frequency of the audiosampling signal by 1/M, for example, 10000/8 times.

Since the phase comparator 1241, the variable crystal oscillator 1242and the frequency divider 1243 form a PLL circuit, the audio clockgeneration unit 1141 can generate an audio sampling signal synchronizedwith the network clock.

As described above, by means of the present embodiment, in addition tothe effect of the third embodiment, an audio sampling signal can begenerated from the network clock by way of the bus clock by the PLLcircuits provided in the bus clock generation unit 1121 and the audioclock generation unit 1141, so that an audio sampling signal of the samefrequency as the network clock which is in phase with the network clockcan be generated.

Embodiment 5

FIG. 13 is a block diagram showing the construction of the PC-basedteleconference terminal 1300 according to the fifth embodiment of thepresent invention. The PC-based teleconference terminal 1300 of thepresent embodiment is principally constructed of the same components asPC-based teleconference terminal 100, and the PC-based teleconferenceterminal 200, so that such components have been given the same referencenumerals, and their explanation has been omitted.

The PC 110, the video codec unit 130, and the audio codec unit 240 areall connected, in the same way as the PC-based teleconference terminal200, to the display unit 150, the camera 151, the microphone 152 and thespeaker 153. The memory 112 again stores the synchronization slipcontrol software 211 and the multiplexer/separator software 212.

The network control unit 1320 is made up of a line switching connectionprocessing unit 121, an ISDN line interface unit 122, a network clockcontrol unit 123 and a line switching transparent data processing unit1321. The line switching transparent data processing unit 1321additionally includes a shared memory 1322.

The shared memory 1322 has an ISDN line interface unit 122 side and acomputer bus 114 side, with simultaneous access of both sides beingpossible, and is a dual port memory fitted with a DMAC (Direct MemoryAccess Controller). The CPU 111 accesses the shared memory 1322 via thecomputer bus 114 in the same way as when accessing the memory 112. Inaddition, the transfer of AV multiframes between the CPU 111 and theISDN line interface unit 122 is executed via this DMAC. In order tostore the transmission queues, the shared memory 1322 is provided withstorage areas, in the same way as shown in FIG. 4 for the firstembodiment, such as a transmission video buffer 404, a transmissionvideo descriptor 400, a transmission audio buffer 440, a transmissionaudio descriptor 430, a transmission buffer 454, and a transmissiondescriptor 450. In the same way, in order to store the reception queues,the shared memory 1322 is provided with storage areas, in the same wayas shown in FIG. 5 for the first embodiment, such as a reception videobuffer 504, a reception video descriptor 500, a reception audio buffer540, a reception audio descriptor 530, a reception buffer 555, and areception descriptor 550.

It should be noted that aside from the line switching transparent dataprocessing unit 1321 and the shared memory 1322, the PC-basedteleconference terminal 1300 is constructed in exactly the same way asthe PC-based teleconference terminal 200 in the first embodiment.Therefore, the details of the operational process of the PC-basedteleconference terminal 1300 have been omitted, and only a basicexplanation will be given.

During reception, the CPU 111, by executing the multiplexer/separatorsoftware 212, queues the received AV multiframes in the reception buffer555 in the shared memory 1322 without transferring to the memory 112,and then setting the storage position of the retrieved frame alignmentsignal (FAS) as the standard, separates the video code 505, the audiocode 541, and the data.

The CPU 111 temporarily queues the separated video code 505 and audiocode 541 respectively in the reception video buffer 504 and in thereception audio buffer 540 in the shared memory 1322. A synchronizationslip control process is executed for the separated audio code 541 by theCPU 111 executing the synchronization slip control software 211.

Following this, a lip synch process is then executed for video code 505and audio code 541 queued in the reception video buffer 504 and in thereception audio buffer 540, before they are transferred to the videocodec circuit 134 and the audio codec circuit 143, respectively.

In the same way, the CPU 111 queues the video code 405 and the audiocode 441 encoded by the video codec circuit 134 and the audio codeccircuit 143 in the transmission video buffer 404 and the transmissionaudio buffer 440 provided in the shared memory 1322, without sendingthem to the memory 112.

A synchronization slip control process is executed for the audio code441 by the CPU 111 executing the synchronization slip control software211.

The CPU 111 then performs a lip-synch process on the video code 405 andthe audio code 441 from the transmission video buffer 404 and thetransmission audio buffer 440 in the transmission buffer 454 provided inthe shared memory 1322, and setting the storage position of the framealignment signal (FAS) as the standard, composes the AV multiframes. Indoing so, the transmission code 455 made up of the AV multiframes arethen queued in the transmission buffer 454 provided in the shared memory1322.

In the manner described above, then, in addition to the effect of thefirst embodiment, by providing a shared memory 1322, equipped with aDMAC, in the line switching transparent data processing unit 1321, theCPU 111, on transmitting or receiving AV multiframes, need not transferthe audio code and video code to the memory 112, so that a saving in theload of the CPU 111 corresponding to this operation can be made.

Although the present invention has been fully described by way ofexamples with reference to the accompanying drawings, it is to be notedthat various changes and modifications will be apparent to those skilledin the art. Therefore, unless such changes and modifications depart fromthe scope of the present invention, they should be construed as beingincluded therein.

What is claimed is:
 1. A teleconference terminal using a personalcomputer for sending and receiving video code, audio code and data toand from another teleconference terminal, using an ISDN line switchingmethod wherein AV multiframes are defined for an H series recommendationfor CCITT, comprising:audio coding means for encoding and compressing aninputted audio signal in accordance with an audio sampling signal, forgenerating transmission audio code and outputting the transmission audiocode to a computer bus, as well as receiving reproduction audio codefrom the computer bus, and generating a reproduction audio signal bydecompressing and decoding the reproduction audio code in accordancewith the audio sampling signal; video coding means for encoding andcompressing an inputted video signal, for generating transmission videocode and outputting the transmission video code to the computer bus, aswell as receiving reproduction video code from the computer bus, andgenerating a reproduction video signal by expanding and decoding thereproduction video code; AV multiframe conversion means for generatingtransmission AV multiframes by performing multiplex conversion of thetransmission audio code and the transmission video code inputted fromthe computer bus according to a CCITT recommendation, and outputting thegenerated transmission AV multiframes to the computer bus; transfermeans for transmitting the transmission AV multiframes inputted from thecomputer bus via an ISDN line to the other teleconference terminal, andfor receiving reproduction AV multiframes from the other teleconferenceterminal according to a CCITT recommendation via the ISDN line, andoutputting the received reproduction AV multiframes to the computer bus;AV separation means for receiving the received reproduction AVmultiframes from the computer bus and for separating the reproductionvideo code and the reproduction audio code from the receivedreproduction AV multiframes, as well as outputting the reproductionvideo code and reproduction audio code to the computer bus; internalclock supplying means for independently generating an internal clock forthe personal computer, as well as supplying the audio sampling signalsynchronized with the internal clock to the audio coding means, whereinthe audio sampling signal has a frequency of 8 kHz; and adjustment meansfor adjusting surpluses and shortages in the transmission audio code tobe included in the transmission AV multiframes, which occur due tosynchronization slips between a network clock of the ISDN line and theinternal clock, and surpluses and shortages in the reproduction audiocode separated from the reproduction AV multiframes, which occur due tosynchronization slips between the network clock of the ISDN line and theinternal clock.
 2. The teleconference terminal of claim 1, wherein theaudio coding means, the video coding means, the transfer means, and theinternal clock supplying means are all constructed on a same expansionboard for the personal computer, and the AV multiframe conversion means,the AV separation means and the adjustment means are all realized byhaving a CPU in the personal computer execute a program stored in amemory.
 3. The teleconference terminal of claim 2, further comprising:atransmission buffer provided in the memory of the personal computer forstoring the transmission AV multiframes; a transmission audio bufferprovided in the memory of the personal computer for storing thetransmission audio code; a reception buffer provided in the memory ofthe personal computer for storing the reproduction AV multiframes; and areception audio buffer provided in the memory of the personal computerfor storing the reproduction audio code, wherein the AV multiframeconversion means includes: a first queuing unit for receiving thetransmission audio code from the computer bus and queuing thetransmission audio code in units of a fixed length in the transmissionaudio buffer; a transmission AV multiframe generation unit for receivinga fixed amount of the transmission audio code from the transmissionaudio buffer, for executing lip-synching, for arranging a framealignment signal (FAS), the transmission video code and the transmissionaudio code, and for generating the transmission AV multiframes in thetransmission buffer according to the CCITT recommendation; and a firsttransfer unit for outputting a fixed amount of the transmission AVmultiframes from the transmission buffer to the computer bus, for eachtime the fixed amount of the transmission AV multiframes are to betransmitted,and the AV separation means includes: a reception framestorage unit for storing the reproduction AV multiframes received fromthe computer bus in the reception buffer; a frame alignment signaldetection unit for detecting a storage position of a frame alignmentsignal (FAS) in the reproduction AV multiframes stored in the receptionbuffer; a frame alignment pointer for storing the storage position ofthe detected frame alignment signal (FAS); an AV separation unit forreceiving a fixed amount of reproduction AV multiframes from thereception buffer, and for separating the reproduction audio code fromthe reproduction AV multiframes, using as a standard the storageposition of the frame alignment signal (FAS) stored in the framealignment pointer; a second queuing unit for queuing the separatedreproduction audio code in units of a fixed amount in the receptionaudio buffer; and a second transfer unit for receiving, each time theaudio coding means generates a fixed amount of the reproduction audiocode, the fixed amount of the reproduction audio code from the receptionaudio buffer and outputting the reproduction audio code to the computerbus.
 4. The teleconference terminal of claim 3, wherein the adjustmentmeans includes:a first speed difference detection unit for detecting aspeed difference between a transfer speed of the transfer means and atransmission audio code generation speed of the audio coding means; afirst stuffing unit for stuffing an appropriate transmission audio codeinto the transmission audio code when the transfer speed is detected asbeing greater than the transmission audio code generation speed; a firstdestuffing unit for removing (destuffing) an appropriate transmissionaudio code from the transmission audio code, when the transfer speed isdetected as being lower than the transmission audio code generationspeed; a second speed difference detection unit for detecting a speeddifference between the transfer speed of the transfer means and areproduction audio signal generation speed of the audio coding means; asecond stuffing unit for stuffing an appropriate reproduction audio codeinto the reproduction audio code when the transfer speed is detected asbeing lower than the reproduction audio code generation speed; and asecond destuffing unit for removing (destuffing) an appropriatereproduction audio code from the reproduction audio code, when thetransfer speed is detected as being greater than the reproduction audiocode generation speed.
 5. The teleconference terminal of claim 4,whereinthe first speed difference detection unit includes: a firstresident amount detection unit for detecting a resident amount of thetransmission audio code in the transmission audio buffer, wherein thefirst speed difference detection unit, by comparing the detectedresident amount of the transmission audio code with a predeterminedresident amount upper limit and a predetermined resident amount lowerlimit, detects the speed difference between the transmission speed ofthe transfer means and the transmission audio code generation speed ofthe audio coding means,wherein the second speed difference detectionunit includes: a second resident amount detection unit for detecting aresident amount of the reproduction audio code in the reception audiobuffer, wherein the second speed difference detection unit, by comparingthe detected resident amount of the reproduction audio code with apredetermined resident amount upper limit and a predetermined residentamount lower limit, detects the speed difference between the receptionspeed of the transfer means and the reproduction audio signal generationspeed of the audio coding means,wherein the adjustment means furtherincludes: a first priority setting unit for setting a predeterminedpriority to each part of a fixed amount in the transmission audio codequeued in the transmission audio buffer; and a second priority settingunit for setting a predetermined priority to each part of the fixedamount in the reproduction audio code queued in the reception audiobuffer, and wherein the first stuffing unit, by having parts of thetransmission audio code set with a lowest priority outputted twice fromthe audio transmission buffer to the transmission AV multiframegeneration unit, stuffs the parts of the transmission audio code; thefirst destuffing unit, by having parts of the transmission audio codeset with a lowest priority outputted from the audio transmission bufferto the transmission AV multiframe generation unit, destuffs the parts ofthe transmission audio code; the second stuffing unit, by having partsof the reproduction audio code set with a lowest priority outputtedtwice from the audio reception buffer to the second transfer unit,stuffs the parts of the reproduction audio code; and the seconddestuffing unit, by not having parts of the reproduction audio code setwith a lowest priority outputted from the audio reproduction buffer tothe second transfer unit and so discarded, destuffs the parts of thereproduction audio code.
 6. The teleconference terminal of claim 5,whereinthe first priority setting unit includes a first sound energydetection unit for calculating, when the audio coding means is encodingusing PCM according to a G.711 recommendation for CCITT, sound energy ineach of the parts of the fixed amount in the transmission audio codequeued in the transmission audio buffer by finding a sum of squares forevery audio sample value, and the first priority setting unit sets thecalculated sound energy as the priority of each of the parts of thefixed amount in the transmission audio code, and the second prioritysetting unit includes a second sound energy detection unit forcalculating, when the audio coding means is encoding using PCM accordingto the G.711 recommendation for CCITT, sound energy in each of the partsof the fixed amount in the reproduction audio code queued in thereception audio buffer by finding a sum of squares for every audiosample value, and the second priority setting unit sets the calculatedsound energy as the priority of each of the parts of the fixed amount inthe reproduction audio code.
 7. The teleconference terminal of claim 5,whereinthe first priority setting unit includes a first addition valuecalculator unit for calculating, when the audio coding means is encodingusing PCM according to a G.726 recommendation for CCITT, an additionvalue for differential information for the parts of the fixed amount inthe transmission audio code queued in the transmission audio buffer, andthe first priority setting unit sets the calculated addition value fordifferential information as the priority of each of the parts of thefixed amount in the transmission audio code, and the second prioritysetting unit includes a second addition value calculator unit forcalculating, when the audio coding means is encoding using PCM accordingto the G.726 recommendation for CCITT, an addition value fordifferential information for the parts of the fixed amount in thereproduction audio code queued in the reception audio buffer, and thesecond priority setting unit sets the calculated addition value fordifferential information as the priority of each of the parts of thefixed amount in the reproduction audio code.
 8. The teleconferenceterminal of claim 5, whereinthe first priority setting unit includes afirst calculator unit for calculating, when the audio coding means isencoding using ADPCM according to a G.726 recommendation for CCITT, asum of squares of differential information for the parts of the fixedamount in the transmission audio code queued in the transmission audiobuffer, and the first priority setting unit sets the calculated sum ofsquares of differential information as the priority of each of the partsof the fixed amount in the transmission audio code, and the secondpriority setting unit includes a second addition value calculator unitfor calculating, when the audio coding means is encoding using ADPCMaccording to the G.726 recommendation for CCITT, a sum of squares ofdifferential information for the parts of the fixed amount in thereproduction audio code queued in the reception audio buffer, and thesecond priority setting unit sets the calculated sum of squares ofdifferential information as the priority of each of the parts of thefixed amount in the reproduction audio code.
 9. The teleconferenceterminal of claim 4, whereinthe first speed difference detection unitincludes:a first resident amount detection unit for detecting a residentamount of the transmission audio code in the transmission audio buffer,wherein the first speed difference detection unit, by comparing thedetected resident amount of the transmission audio code with apredetermined resident amount upper limit and a predetermined residentamount lower limit, detects the speed difference between thetransmission speed of the transfer means and the transmission audio codegeneration speed of the audio coding means,wherein the second speeddifference detection unit includes: a second resident amount detectionunit for detecting a resident amount of the reproduction audio code inthe reception audio buffer, wherein the second speed differencedetection unit, by comparing the detected resident amount of thereproduction audio code with a predetermined resident amount upper limitand a predetermined resident amount lower limit, detects the speeddifference between the reception speed of the transfer means and thereproduction audio signal generation speed of the audio codingmeans,wherein the adjustment means further includes: a first prioritysetting unit for setting a predetermined priority to each part of thefixed amount in the transmission audio code queued in the transmissionaudio buffer; a second priority setting unit for setting a predeterminedpriority to each part of the fixed amount in the reproduction audio codequeued in the reception audio buffer; a transmission interpolation audiocode generation unit for calculating average values of audio samplevalues for at least two out of a part of the audio transmission codewith a lowest priority, a part of the audio transmission code queued soas to follow after the part of the audio transmission code with thelowest priority, and a part of the audio transmission code queued so asto precede the part of the audio transmission code with the lowestpriority, and for generating transmission interpolation audio code withthe calculated average values as audio sample values; and a reproductioninterpolation audio code generation unit for calculating average valuesof audio sample values for at least two out of a part of the audioreproduction code with a lowest priority, a part of the audioreproduction code queued so as to follow after the part of the audioreproduction code with the lowest priority, and a part of the audioreproduction code queued so as to precede the part of the audioreproduction code with the lowest priority, and for generatingreproduction interpolation audio code with the calculated average valuesas audio sample values,wherein the first stuffing unit, by having thetransmission interpolation audio code outputted following a part of thetransmission audio code set with a lowest priority in the audiotransmission buffer to the transmission AV multiframe generation unit,stuffs the transmission interpolation audio code; the first destuffingunit, by having the transmission interpolation audio code outputted tothe transmission AV multiframe generation unit in place of the part ofthe transmission audio code set with a lowest priority and the part ofthe audio transmission information queued so as to follow directly afterthe part of the audio transmission code with the lowest priority in theaudio transmission buffer, destuffs the part of the transmission audiocode with the lowest priority; the second stuffing unit, by having thereproduction interpolation audio code outputted following a part of thereproduction audio code set with a lowest priority from the audioreception buffer to the second transfer unit, stuffs the reproductioninterpolation audio code; and the second destuffing unit, by having thereproduction interpolation audio code outputted to the second transferunit in place of the part of the reproduction audio code set with alowest priority and the part of the audio reproduction informationqueued so as to follow directly after the part of the audio reproductioncode with the lowest priority in the audio reception buffer, destuffsthe part of the reproduction audio code with the lowest priority. 10.The teleconference terminal of claim 9, whereinthe first prioritysetting unit includes a first sound energy detection unit forcalculating, when the audio coding means is encoding using PCM accordingto a G.711 recommendation for CCITT, sound energy in each of the partsof the fixed amount in the transmission audio code queued in thetransmission audio buffer by finding a sum of squares for every audiosample value, and the first priority setting unit sets the calculatedsound energy as the priority of each of the parts of the fixed amount inthe transmission audio code, and the second priority setting unitincludes a second sound energy detection unit for calculating, when theaudio coding means is encoding using PCM according to the G.711recommendation for CCITT, sound energy in each of the parts of the fixedamount in the reproduction audio code queued in the reception audiobuffer by finding a sum of squares for every audio sample value, and thesecond priority setting unit sets the calculated sound energy as thepriority of each of the parts of the fixed amount in the reproductionaudio code.
 11. The teleconference terminal of claim 9, whereinthe firstpriority setting unit includes a first addition value calculator unitfor calculating, when the audio coding means is encoding using PCMaccording to a G.726 recommendation for CCITT, an addition value fordifferential information for the parts of the fixed amount in thetransmission audio code queued in the transmission audio buffer, and thefirst priority setting unit sets the calculated addition value fordifferential information as the priority of each of the parts of thefixed amount in the transmission audio code, and the second prioritysetting unit includes a second addition value calculator unit forcalculating, when the audio coding means is encoding using PCM accordingto the G.726 recommendation for CCITT, an addition value fordifferential information for the parts of the fixed amount in thereproduction audio code queued in the reception audio buffer, and thesecond priority setting unit sets the calculated addition value fordifferential information as the priority of each of the parts of thefixed amount in the reproduction audio code.
 12. The teleconferenceterminal of claim 9, whereinthe first priority setting unit includes afirst calculator unit for calculating, when the audio coding means isencoding using ADPCM according to a G.726 recommendation for CCITT, asum of squares of differential information for the parts of the fixedamount in the transmission audio code queued in the transmission audiobuffer, and the first priority setting unit sets the calculated sum ofsquares of differential information as the priority of each of the partsof the fixed amount in the transmission audio code, and the secondpriority setting unit includes a second addition value calculator unitfor calculating, when the audio coding means is encoding using ADPCMaccording to the G.726 recommendation for CCITT, a sum of squares ofdifferential information for the parts of the fixed amount in thereproduction audio code queued in the reception audio buffer, and thesecond priority setting unit sets the calculated sum of squares ofdifferential information as the priority of each of the parts of thefixed amount in the reproduction audio code.
 13. The teleconferenceterminal of claim 3, whereinthe transfer means includes: an ISDN lineinterface unit for transmitting the transmission AV multiframes on theISDN line and for receiving the reproduction AV multiframes from theISDN line; a shared memory which is a dual port memory which can besimultaneously accessed by the CPU in the personal computer and the ISDNline interface unit; and a DMAC for directly storing the receivedreproduction AV multiframes in the shared memory and for transferringthe transmission AV multiframes stored in the shared memory to the ISDNline interface unit.
 14. The teleconference terminal of claim 13,whereinthe adjustment means includes:a first speed difference detectionunit for detecting a speed difference between a transfer speed of thetransfer means and a transmission audio code generation speed of theaudio coding means; a first stuffing unit for stuffing an appropriatetransmission audio code into the transmission audio code when thetransfer speed is detected as being greater than the transmission audiocode generation speed; a first destuffing unit for removing (destuffing)an appropriate transmission audio code from the transmission audio code,when the transfer speed is detected as being lower than the transmissionaudio code generation speed; a second speed difference detection unitfor detecting a speed difference between the transfer speed of thetransfer means and a reproduction audio signal generation speed of theaudio coding means; a second stuffing unit for stuffing an appropriatereproduction audio code into the reproduction audio code when thetransfer speed is detected as being lower than the reproduction audiocode generation speed; and a second destuffing unit for removing(destuffing) an appropriate reproduction audio code from thereproduction audio code, when the transfer speed is detected as beinggreater than the reproduction audio code generation speed.
 15. Theteleconference terminal of claim 14, whereinthe first speed differencedetection unit includes:a first resident amount detection unit fordetecting a resident amount of the transmission audio code in thetransmission audio buffer, wherein the first speed difference detectionunit, by comparing the detected resident amount of the transmissionaudio code with a predetermined resident amount upper limit and apredetermined resident amount lower limit, detects the speed differencebetween the transmission speed of the transfer means and thetransmission audio code generation speed of the audio codingmeans,wherein the second speed difference detection unit includes: asecond resident amount detection unit for detecting a resident amount ofthe reproduction audio code in the reception audio buffer, wherein thesecond speed difference detection unit, by comparing the detectedresident amount of the reproduction audio code with a predeterminedresident amount upper limit and a predetermined resident amount lowerlimit, detects the speed difference between the reception speed of thetransfer means and the reproduction audio signal generation speed of theaudio coding means,wherein the adjustment means further includes: afirst priority setting unit for setting a predetermined priority to eachpart of a fixed amount in the transmission audio code queued in thetransmission audio buffer; and a second priority setting unit forsetting a predetermined priority to each part of the fixed amount in thereproduction audio code queued in the reception audio buffer, andwherein the first stuffing unit, by having parts of the transmissionaudio code set with a lowest priority outputted twice from the audiotransmission buffer to the transmission AV multiframe generation unit,stuffs the parts of the transmission audio code; the first destuffingunit, by having parts of the transmission audio code set with a lowestpriority outputted from the audio transmission buffer to thetransmission AV multiframe generation unit, destuffs the parts of thetransmission audio code; the second stuffing unit, by having parts ofthe reproduction audio code set with a lowest priority outputted twicefrom the audio reception buffer to the second transfer unit, stuffs theparts of the reproduction audio code; and the second destuffing unit, bynot having parts of the reproduction audio code set with a lowestpriority outputted from the audio reproduction buffer to the secondtransfer unit and so discarded, destuffs the parts of the reproductionaudio code.
 16. The teleconference terminal of claim 15, whereinthefirst priority setting unit includes a first sound energy detection unitfor calculating, when the audio coding means is encoding using PCMaccording to a G.711 recommendation for CCITT, sound energy in each ofthe parts of the fixed amount in the transmission audio code queued inthe transmission audio buffer by finding a sum of squares for everyaudio sample value, and the first priority setting unit sets thecalculated sound energy as the priority of each of the parts of thefixed amount in the transmission audio code, and the second prioritysetting unit includes a second sound energy detection unit forcalculating, when the audio coding means is encoding using PCM accordingto the G.711 recommendation for CCITT, sound energy in each of the partsof the fixed amount in the reproduction audio code queued in thereception audio buffer by finding a sum of squares for every audiosample value, and the second priority setting unit sets the calculatedsound energy as the priority of each of the parts of the fixed amount inthe reproduction audio code.
 17. The teleconference terminal of claim15, whereinthe first priority setting unit includes a first additionvalue calculator unit for calculating, when the audio coding means isencoding using PCM according to a G.726 recommendation for CCITT, anaddition value for differential information for the parts of the fixedamount in the transmission audio code queued in the transmission audiobuffer, and the first priority setting unit sets the calculated additionvalue for differential information as the priority of each of the partsof the fixed amount in the transmission audio code, and the secondpriority setting unit includes a second addition value calculator unitfor calculating, when the audio coding means is encoding using PCMaccording to the G.726 recommendation for CCITT, an addition value fordifferential information for the parts of the fixed amount in thereproduction audio code queued in the reception audio buffer, and thesecond priority setting unit sets the calculated addition value fordifferential information as the priority of each of the parts of thefixed amount in the reproduction audio code.
 18. The teleconferenceterminal of claim 15, whereinthe first priority setting unit includes afirst calculator unit for calculating, when the audio coding means isencoding using ADPCM according to a G.726 recommendation for CCITT, asum of squares of differential information for the parts of the fixedamount in the transmission audio code queued in the transmission audiobuffer, and the first priority setting unit sets the calculated sum ofsquares of differential information as the priority of each of the partsof the fixed amount in the transmission audio code, and the secondpriority setting unit includes a second addition value calculator unitfor calculating, when the audio coding means is encoding using ADPCMaccording to the G.726 recommendation for CCITT, a sum of squares ofdifferential information for the parts of the fixed amount in thereproduction audio code queued in the reception audio buffer, and thesecond priority setting unit sets the calculated sum of squares ofdifferential information as the priority of each of the parts of thefixed amount in the reproduction audio code.
 19. The teleconferenceterminal of claim 2, which does not comprise the internal clocksupplying means and the adjustment means, wherein the transfer meansincludes a clock distribution circuit for distributing the network clockof the ISDN line, and a special connector is formed in each of thetransfer means and the audio coding means, with a private clock signalline connected between the special connectors, so that the network clockdistributed by the clock distribution circuit is supplied via theprivate clock signal line to the audio coding means.
 20. Theteleconference terminal of claim 2, comprising in place of the internalclock supplying means and the adjustment means:a count-up pulsegeneration unit for generating a same number of pulses as a number ofaudio samples included in a fixed amount of the reproduction audio codeinputted at a time by the audio coding means from the computer bus; avoltage control oscillator for outputting a clock signal of a frequencyset in accordance with an applied voltage; an updown counter forinputting the clock signal from the voltage control oscillator as acount-down signal and inputting the pulses from the count-up pulsegeneration unit as a count-up signal, and for outputting a differencebetween the count-up signal and the count-down signal; and an integratorfor smoothing an output of the updown counter by integrating, andoutputting to the voltage control oscillator, wherein an output of thevoltage control oscillator is supplied to the audio coding means as theaudio sampling signal.
 21. The teleconference terminal of claim 20,wherein the count-up pulse generation unit includes:an I/O decoder fordecoding a transfer address of the fixed amount of the reproductionaudio code transferred via the computer bus, and, when the transferaddress is the audio coding means, for generating a write pulse forwriting data in the reproduction audio code into the audio coding means,as well as generating a register write pulse; an audio setting registerfor retrieving a number of audio samples written at a start of thereproduction audio code in accordance with the register write pulse andfor storing the number of audio samples; a flip flop which is setaccording to the write pulse for the audio coding means and reset by aclear signal; a counter for counting up a number of clock pulsesinputted while the flip flop is set, and for being reset to 1! when theflip flop is reset; an agreement circuit for outputting the clear signalto the flip flop when a count value of the counter is equal to thenumber of audio samples stored by the audio setting register; an ANDcircuit for outputting the clock pulses as the count-up signal while theflip flop is set; and a clock oscillator for supplying the clock pulsesto the flip flop, the counter and the AND gate.
 22. The teleconferenceterminal of claim 2, comprising in place of the internal clock supplyingmeans and the adjustment means:a bus clock generation unit formultiplying the network clock from the transfer means and generating abus clock for the computer bus synchronized to the network clock; and anaudio sampling signal generation unit generation unit for dividing thebus clock, for generating the audio sampling signal synchronized to thebus clock and supplying the audio sampling signal to the audio codingmeans.
 23. The teleconference terminal of claim 1, comprising in placeof the internal clock supplying means and the adjustment means:acount-up pulse generation unit for generating a same number of pulses asa number of audio samples included in a fixed amount of the reproductionaudio code inputted at a time by the audio coding means from thecomputer bus; a voltage control oscillator for outputting a clock signalof a frequency set in accordance with an applied voltage; an updowncounter for inputting the clock signal from the voltage controloscillator as a count-down signal and inputting the pulses from thecount-up pulse generation unit as a count-up signal, and for outputtinga difference between tie count-up signal and the count-down signal; andan integrator for smoothing an output of the updown counter byintegrating, and outputting to the voltage control oscillator, whereinan output of the voltage control oscillator is supplied to the audiocoding means as the audio sampling signal.
 24. The teleconferenceterminal of claim 1, wherein the count-up pulse generation unitincludes:an I/O decoder for decoding a transfer address of the fixedamount of the reproduction audio code transferred via the computer bus,and, when the transfer address is the audio coding means, for generatinga write pulse for writing data in the reproduction audio code into theaudio coding means, as well as generating a register write pulse; anaudio setting register for retrieving a number of audio samples writtenat a start of the reproduction audio code in accordance with theregister write pulse and for storing the number of audio samples; a flipflop which is set according to the write pulse for the audio codingmeans and reset by a clear signal; a counter for counting up a number ofclock pulses inputted while the flip flop is set, and for being reset to1! when the flip flop is reset; an agreement circuit for outputting theclear signal to the flip flop when a count value of the counter is equalto the number of audio samples stored by the audio setting register; anAND circuit for outputting the clock pulses as the count-up signal whilethe flip flop is set; and a clock oscillator for supplying the clockpulses to the flip flop, the counter and the AND gate.
 25. Theteleconference terminal of claim 1, wherein comprising in place of theinternal clock supplying means and the adjustment means:a bus clockgeneration unit for multiplying the network clock from the transfermeans and generating a bus clock for the computer bus synchronized tothe network clock; and an audio sampling signal generation unitgeneration unit for dividing the bus clock, for generating the audiosampling signal synchronized to the bus clock and supplying the audiosampling signal to the audio coding means.